Linked Questions

6
votes
6answers
1k views

Why can't a causal digital filter have an infinitely sharp transition between the passband and the stopband?

In DSP book by Proakis and as well as in this pdf, it is mentioned that practical causal digital filters cannot have an infinitely sharp transition from Pass-band to Stop-band. Why is it so? Can you ...
3
votes
2answers
44k views

Removing noise from audio using Fourier transform in Matlab

I'm trying to remove noise from an audio file. This audio file contains speech as well as constant pink noise. I know that I have to use the Fourier transform to convert to the frequency domain then ...
15
votes
2answers
7k views

A good mathematical explanation of Gibbs phenomenon

I was explaining to someone how Fourier series work in context of constructing signals that are not everywhere differentiable, e.g. square waves, sawtooth waves, etc. When I mentioned the Gibbs ...
3
votes
2answers
4k views

Low pass filter with DFT

This is my first question at this site, please excuse me if I'm not asking the question properly. I have an acoustic signal of a test and I want to remove the "thin" peaks, because they don't really ...
4
votes
3answers
889 views

Chop out frequencies outside human hearing range

I have a bunch of audio files all sampled at 44100 Hz sample frequency. I am trying to remove all the frequencies which are outside the human hearing range (I use the following as reference: Frequency ...
2
votes
1answer
2k views

simplest Notch filter in the world

What do you think of this notch / band reject filter for a very very narrow band ? ...
1
vote
3answers
3k views

FFT of any sinusoidal signal — how many point fft to take?

suppose I have to measure the DFT of an 60 hz sinusoidal signal and calculate upto 20 harmonics of this. Value of the 20th harmonic will be 20*60 = 1200 HZ. Then my sampling frequency should be >= 20*...
5
votes
2answers
2k views

Zeroing bins before IFFT

Endolith's recent question on zeroing FFT bins got me thinking about a possible error in my workflow and I'd like to double check. Consider a signal $x_i[m],\ m\in\{1,\ldots,M\}$, i.e. $M$ samples ...
0
votes
2answers
2k views

How to interpret results of Fourier transform by using Python

Connected with this topic , I would like to have some clarifications about the Fourier Transform and the Python tools that make it. Having my data set whose plot is showed in the above link, my array ...
0
votes
2answers
2k views

High-pass filtering pre-processing before computing audio features

I want to compute acoustic features on a set of audio files. These features can be classical audio features as spectral flux, but also acoustic indices as proposed in the seewave R package ). I need,...
0
votes
2answers
3k views

What's a “Fourier filter”?

E.g. the constant Q-transform is built by adding so called "Fourier filters". What's a "Fourier filter"?
1
vote
1answer
3k views

FFT on non-periodic signal and signal power at each harmonic - is there a more robust approach?

I have a non-periodic signal collected from a force plate, it represents the foot/ground contact of a human when running. My aim is to calculate the signal power of each harmonic up to 50Hz. I will ...
4
votes
4answers
1k views

Removing cyclic noise from signal

I'm using a sensor with output that looks like the following figure. The cyclic noise is common to ask measurements made by the sensor. Is there a way to remove the noise without negatively effecting ...
5
votes
1answer
2k views

How to improve an FFT AM demodulator for NOAA APT satellite imagery?

My question involves proper FFT Technique for AM demodulation, in the special case where the modulation rate exceeds the carrier frequency. This question has a specific hobbyist application, the ...
0
votes
1answer
2k views

Simple FFT filtering vs. e.g. butterworth filtering

I am currently working on functional connectivity analysis of EEG, and need to bandpass filter my data into different frequency bands (Delta, Theta, Alpha and Beta). An important thing is that the ...

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